static void _v_readstring(ref Ogg.oggpack_buffer o, ref char[] buf, int bytes)
{
for (int i = 0; i < bytes; i++)
{
buf[i] = (char)Ogg.oggpack_read(ref o, 8);
}
}
/* pack side */
static public int _vorbis_pack_info(ref Ogg.oggpack_buffer opb, ref vorbis_info vi)
{
codec_setup_info ci = vi.codec_setup as codec_setup_info;
if (ci == null)
{
return(OV_EFAULT);
}
/* preamble */
Ogg.oggpack_write(ref opb, 0x01, 8);
_v_writestring(ref opb, new char[] { 'v', 'o', 'r', 'b', 'i', 's' }, 6);
/* basic information about the stream */
Ogg.oggpack_write(ref opb, 0x00, 32);
Ogg.oggpack_write(ref opb, (uint)vi.channels, 8);
Ogg.oggpack_write(ref opb, (uint)vi.rate, 32);
Ogg.oggpack_write(ref opb, (uint)vi.bitrate_upper, 32);
Ogg.oggpack_write(ref opb, (uint)vi.bitrate_nominal, 32);
Ogg.oggpack_write(ref opb, (uint)vi.bitrate_lower, 32);
Ogg.oggpack_write(ref opb, (uint)ilog2((uint)ci.blocksizes[0]), 4);
Ogg.oggpack_write(ref opb, (uint)ilog2((uint)ci.blocksizes[1]), 4);
Ogg.oggpack_write(ref opb, 1, 1);
return(0);
}
/* helpers */
static void _v_writestring(ref Ogg.oggpack_buffer o, char[] s, int bytes)
{
for (int i = 0; i < bytes; i++)
{
Ogg.oggpack_write(ref o, s[i], 8);
}
}
/* used to track pcm position without actually performing decode. Useful for sequential 'fast forward' */
static public int vorbis_synthesis_trackonly(ref vorbis_block vb, ref Ogg.ogg_packet op)
{
vorbis_dsp_state vd = vb.vd;
private_state b = vd.backend_state as private_state;
vorbis_info vi = vd.vi;
codec_setup_info ci = vi.codec_setup as codec_setup_info;
Ogg.oggpack_buffer opb = vb.opb;
int mode;
/* first things first. Make sure decode is ready */
_vorbis_block_ripcord(ref vb);
Ogg.oggpack_readinit(ref opb, op.packet, op.bytes);
/* Check the packet type */
if (Ogg.oggpack_read(ref opb, 1) != 0)
{
/* Oops. This is not an audio data packet */
return(OV_ENOTAUDIO);
}
/* read our mode and pre/post windowsize */
mode = Ogg.oggpack_read(ref opb, b.modebits);
if (mode == -1)
{
return(OV_EBADPACKET);
}
vb.mode = mode;
vb.W = ci.mode_param[mode].blockflag;
if (vb.W > 0)
{
vb.lW = Ogg.oggpack_read(ref opb, 1);
vb.nW = Ogg.oggpack_read(ref opb, 1);
if (vb.nW == -1)
{
return(OV_EBADPACKET);
}
}
else
{
vb.lW = 0;
vb.nW = 0;
}
/* more setup */
vb.granulepos = op.granulepos;
vb.sequence = op.packetno;
vb.eofflag = op.e_o_s;
/* no pcm */
vb.pcmend = 0;
vb.pcm = null;
return(0);
}
static void mapping0_pack(ref vorbis_info vi, vorbis_info_mapping vm, ref Ogg.oggpack_buffer opb)
{
int i;
vorbis_info_mapping0 info = vm as vorbis_info_mapping0;
/* another 'we meant to do it this way' hack... up to beta 4, we packed 4 binary zeros here to signify one submapping in use. We
* now redefine that to mean four bitflags that indicate use of deeper features; bit0:submappings, bit1:coupling,
* bit2,3: reserved. This is backward compatable with all actual uses of the beta code. */
if (info.submaps > 1)
{
Ogg.oggpack_write(ref opb, 1, 1);
Ogg.oggpack_write(ref opb, (uint)(info.submaps - 1), 4);
}
else
{
Ogg.oggpack_write(ref opb, 0, 1);
}
if (info.coupling_steps > 0)
{
Ogg.oggpack_write(ref opb, 1, 1);
Ogg.oggpack_write(ref opb, (uint)(info.coupling_steps - 1), 8);
for (i = 0; i < info.coupling_steps; i++)
{
Ogg.oggpack_write(ref opb, (uint)info.coupling_mag[i], ilog((uint)vi.channels));
Ogg.oggpack_write(ref opb, (uint)info.coupling_ang[i], ilog((uint)vi.channels));
}
}
else
{
Ogg.oggpack_write(ref opb, 0, 1);
}
Ogg.oggpack_write(ref opb, 0, 2); /* 2,3:reserved */
/* we don't write the channel submappings if we only have one... */
if (info.submaps > 1)
{
for (i = 0; i < vi.channels; i++)
{
Ogg.oggpack_write(ref opb, (uint)info.chmuxlist[i], 4);
}
}
for (i = 0; i < info.submaps; i++)
{
Ogg.oggpack_write(ref opb, 0, 8); /* time submap unused */
Ogg.oggpack_write(ref opb, (uint)info.floorsubmap[i], 8);
Ogg.oggpack_write(ref opb, (uint)info.residuesubmap[i], 8);
}
}
/* Is this packet a vorbis ID header? */
static public int vorbis_synthesis_idheader(ref Ogg.ogg_packet op)
{
Ogg.oggpack_buffer opb = new Ogg.oggpack_buffer();
char[] buffer = new char[6];
if (op == null)
{
return(0);
}
Ogg.oggpack_readinit(ref opb, op.packet, op.bytes);
if (op.b_o_s == 0)
{
return(0); /* Not the initial packet */
}
if (Ogg.oggpack_read(ref opb, 8) != 1)
{
return(0); /* Not an ID header */
}
_v_readstring(ref opb, ref buffer, 6);
if
(
buffer[0] != 'v' ||
buffer[1] != 'o' ||
buffer[2] != 'r' ||
buffer[3] != 'b' ||
buffer[4] != 'i' ||
buffer[5] != 's'
)
{
return(0); /* not vorbis */
}
return(1);
}
static public int vorbis_packet_blocksize(ref vorbis_info vi, ref Ogg.ogg_packet op)
{
codec_setup_info ci = vi.codec_setup as codec_setup_info;
Ogg.oggpack_buffer opb = new Ogg.oggpack_buffer();
int mode;
Ogg.oggpack_readinit(ref opb, op.packet, op.bytes);
/* Check the packet type */
if (Ogg.oggpack_read(ref opb, 1) != 0)
{
/* Oops. This is not an audio data packet */
return(OV_ENOTAUDIO);
}
{
int modebits = 0;
int v = ci.modes;
while (v > 1)
{
modebits++;
v >>= 1;
}
/* read our mode and pre/post windowsize */
mode = Ogg.oggpack_read(ref opb, modebits);
}
if (mode == -1)
{
return(OV_EBADPACKET);
}
return(ci.blocksizes[ci.mode_param[mode].blockflag]);
}
static public int vorbis_synthesis(ref vorbis_block vb, ref Ogg.ogg_packet op)
{
vorbis_dsp_state vd = (vb != null) ? vb.vd : null;
private_state b = (vd != null) ? (vd.backend_state as private_state) : null;
vorbis_info vi = (vd != null) ? vd.vi : null;
codec_setup_info ci = (vi != null) ? (vi.codec_setup as codec_setup_info) : null;
Ogg.oggpack_buffer opb = (vb != null) ? vb.opb : null;
int type, mode, i;
if (vd == null || b == null || vi == null || ci == null || opb == null)
{
return(OV_EBADHEADER);
}
/* first things first. Make sure decode is ready */
_vorbis_block_ripcord(ref vb);
Ogg.oggpack_readinit(ref opb, op.packet, op.bytes);
/* Check the packet type */
if (Ogg.oggpack_read(ref opb, 1) != 0)
{
/* Oops. This is not an audio data packet */
return(OV_ENOTAUDIO);
}
/* read our mode and pre/post windowsize */
mode = Ogg.oggpack_read(ref opb, b.modebits);
if (mode == -1)
{
return(OV_EBADPACKET);
}
vb.mode = mode;
if (ci.mode_param[mode] == null)
{
return(OV_EBADPACKET);
}
vb.W = ci.mode_param[mode].blockflag;
if (vb.W > 0)
{
/* this doesn't get mapped through mode selection as it's used only for window selection */
vb.lW = Ogg.oggpack_read(ref opb, 1);
vb.nW = Ogg.oggpack_read(ref opb, 1);
if (vb.nW == -1)
{
return(OV_EBADPACKET);
}
}
else
{
vb.lW = 0;
vb.nW = 0;
}
/* more setup */
vb.granulepos = op.granulepos;
vb.sequence = op.packetno;
vb.eofflag = op.e_o_s;
/* alloc pcm passback storage */
vb.pcmend = ci.blocksizes[vb.W];
vb.pcm = (float **)_vorbis_block_alloc(ref vb, sizeof(float *) * vi.channels);
for (i = 0; i < vi.channels; i++)
{
vb.pcm[i] = (float *)_vorbis_block_alloc(ref vb, sizeof(float) * vb.pcmend);
}
/* unpack_header enforces range checking */
type = ci.map_type[ci.mode_param[mode].mapping];
return(_mapping_P[type].inverse(ref vb, ci.map_param[ci.mode_param[mode].mapping]));
}
/* The Vorbis header is in three packets; the initial small packet in the first page that identifies basic parameters, a second packet
* with bitstream comments and a third packet that holds the codebook. */
static public int vorbis_synthesis_headerin(ref vorbis_info vi, ref vorbis_comment vc, ref Ogg.ogg_packet op)
{
Ogg.oggpack_buffer opb = new Ogg.oggpack_buffer();
Ogg.oggpack_readinit(ref opb, op.packet, op.bytes);
/* Which of the three types of header is this? */
/* Also verify header-ness, vorbis */
{
char[] buffer = new char[6];
int packtype = Ogg.oggpack_read(ref opb, 8);
_v_readstring(ref opb, ref buffer, 6);
if
(
buffer[0] != 'v' ||
buffer[1] != 'o' ||
buffer[2] != 'r' ||
buffer[3] != 'b' ||
buffer[4] != 'i' ||
buffer[5] != 's'
)
{
/* not a vorbis header */
return(OV_ENOTVORBIS);
}
switch (packtype)
{
case 0x01: /* least significant *bit* is read first */
{
if (op.b_o_s == 0)
{
/* Not the initial packet */
return(OV_EBADHEADER);
}
if (vi.rate != 0)
{
/* previously initialized info header */
return(OV_EBADHEADER);
}
}
return(_vorbis_unpack_info(ref vi, ref opb));
case 0x03: /* least significant *bit* is read first */
{
if (vi.rate == 0)
{
/* um... we didn't get the initial header */
return(OV_EBADHEADER);
}
}
return(_vorbis_unpack_comment(ref vc, ref opb));
case 0x05: /* least significant *bit* is read first */
{
if (vi.rate == 0 || vc.vendor == null)
{
/* um... we didn;t get the initial header or comments yet */
return(OV_EBADHEADER);
}
}
return(_vorbis_unpack_books(ref vi, ref opb));
default:
{
/* Not a valid vorbis header type */
return(OV_EBADHEADER);
}
}
}
}
/* all of the real encoding details are here. The modes, books, everything */
static public int _vorbis_unpack_books(ref vorbis_info vi, ref Ogg.oggpack_buffer opb)
{
codec_setup_info ci = vi.codec_setup as codec_setup_info;
if (ci == null)
{
return(OV_EFAULT);
}
/* code books */
ci.books = Ogg.oggpack_read(ref opb, 8) + 1;
if (ci.books <= 0)
{
goto err_out;
}
for (int i = 0; i < ci.books; i++)
{
ci.book_param[i] = vorbis_staticbook_unpack(ref opb);
if (ci.book_param[i] == null)
{
goto err_out;
}
}
/* time backend settings; hooks are unused */
{
int times = Ogg.oggpack_read(ref opb, 6) + 1;
if (times <= 0)
{
goto err_out;
}
for (int i = 0; i < times; i++)
{
int test = Ogg.oggpack_read(ref opb, 16);
if (test < 0 || test >= VI_TIMEB)
{
goto err_out;
}
}
}
/* floor backend settings */
ci.floors = Ogg.oggpack_read(ref opb, 6) + 1;
if (ci.floors <= 0)
{
goto err_out;
}
for (int i = 0; i < ci.floors; i++)
{
ci.floor_type[i] = Ogg.oggpack_read(ref opb, 16);
if (ci.floor_type[i] < 0 || ci.floor_type[i] >= VI_FLOORB)
{
goto err_out;
}
ci.floor_param[i] = _floor_P[ci.floor_type[i]].unpack(ref vi, ref opb);
if (ci.floor_param[i] == null)
{
goto err_out;
}
}
/* residue backend settings */
ci.residues = Ogg.oggpack_read(ref opb, 6) + 1;
if (ci.residues <= 0)
{
goto err_out;
}
for (int i = 0; i < ci.residues; i++)
{
ci.residue_type[i] = Ogg.oggpack_read(ref opb, 16);
if (ci.residue_type[i] < 0 || ci.residue_type[i] >= VI_RESB)
{
goto err_out;
}
ci.residue_param[i] = _residue_P[ci.residue_type[i]].unpack(ref vi, ref opb);
if (ci.residue_param == null)
{
goto err_out;
}
}
/* map backed settings */
ci.maps = Ogg.oggpack_read(ref opb, 6) + 1;
if (ci.maps <= 0)
{
goto err_out;
}
for (int i = 0; i < ci.maps; i++)
{
ci.map_type[i] = Ogg.oggpack_read(ref opb, 16);
if (ci.map_type[i] < 0 || ci.map_type[i] >= VI_MAPB)
{
goto err_out;
}
ci.map_param[i] = _mapping_P[ci.map_type[i]].unpack(ref vi, ref opb);
if (ci.map_param[i] == null)
{
goto err_out;
}
}
/* mode settings */
ci.modes = Ogg.oggpack_read(ref opb, 6) + 1;
if (ci.modes <= 0)
{
goto err_out;
}
for (int i = 0; i < ci.modes; i++)
{
ci.mode_param[i] = new vorbis_info_mode();
ci.mode_param[i].blockflag = Ogg.oggpack_read(ref opb, 1);
ci.mode_param[i].windowtype = Ogg.oggpack_read(ref opb, 16);
ci.mode_param[i].transformtype = Ogg.oggpack_read(ref opb, 16);
ci.mode_param[i].mapping = Ogg.oggpack_read(ref opb, 8);
if (ci.mode_param[i].windowtype >= VI_WINDOWB)
{
goto err_out;
}
if (ci.mode_param[i].transformtype >= VI_WINDOWB)
{
goto err_out;
}
if (ci.mode_param[i].mapping >= ci.maps)
{
goto err_out;
}
if (ci.mode_param[i].mapping < 0)
{
goto err_out;
}
}
if (Ogg.oggpack_read(ref opb, 1) != 1)
{
goto err_out;
}
return(0);
err_out:
vorbis_info_clear(ref vi);
return(OV_EBADHEADER);
}
static int _vorbis_unpack_comment(ref vorbis_comment vc, ref Ogg.oggpack_buffer opb)
{
int i;
int vendorlen = Ogg.oggpack_read(ref opb, 32);
if (vendorlen < 0)
{
goto err_out;
}
if (vendorlen > opb.storage - 8)
{
goto err_out;
}
vc.vendor = new char[vendorlen + 1];
_v_readstring(ref opb, ref vc.vendor, vendorlen);
i = Ogg.oggpack_read(ref opb, 32);
if (i < 0)
{
goto err_out;
}
if (i > ((opb.storage - Ogg.oggpack_bytes(ref opb) >> 2)))
{
goto err_out;
}
vc.comments = i;
vc.user_comments = new char[vc.comments + 1][];
vc.comment_lengths = new int[vc.comments + 1];
for (i = 0; i < vc.comments; i++)
{
int len = Ogg.oggpack_read(ref opb, 32);
if (len < 0)
{
goto err_out;
}
if (len > opb.storage - Ogg.oggpack_bytes(ref opb))
{
goto err_out;
}
vc.comment_lengths[i] = len;
vc.user_comments[i] = new char[len + 1];
_v_readstring(ref opb, ref vc.user_comments[i], len);
}
/* EOP check */
if (Ogg.oggpack_read(ref opb, 1) != 1)
{
goto err_out;
}
return(0);
err_out:
vorbis_comment_clear(ref vc);
return(OV_EBADHEADER);
}
/* Header packing/unpacking */
static public int _vorbis_unpack_info(ref vorbis_info vi, ref Ogg.oggpack_buffer opb)
{
codec_setup_info ci = vi.codec_setup as codec_setup_info;
if (ci == null)
{
return(OV_EFAULT);
}
vi.version = Ogg.oggpack_read(ref opb, 32);
if (vi.version != 0)
{
return(OV_EVERSION);
}
vi.channels = Ogg.oggpack_read(ref opb, 8);
vi.rate = Ogg.oggpack_read(ref opb, 32);
vi.bitrate_upper = Ogg.oggpack_read(ref opb, 32);
vi.bitrate_nominal = Ogg.oggpack_read(ref opb, 32);
vi.bitrate_lower = Ogg.oggpack_read(ref opb, 32);
ci.blocksizes[0] = 1 << Ogg.oggpack_read(ref opb, 4);
ci.blocksizes[1] = 1 << Ogg.oggpack_read(ref opb, 4);
if (vi.rate < 1)
{
goto err_out;
}
if (vi.channels < 1)
{
goto err_out;
}
if (ci.blocksizes[0] < 64)
{
goto err_out;
}
if (ci.blocksizes[1] < ci.blocksizes[0])
{
goto err_out;
}
if (ci.blocksizes[1] > 8192)
{
goto err_out;
}
if (Ogg.oggpack_read(ref opb, 1) != 1)
{
goto err_out; /* EOP check */
}
return(0);
err_out:
vorbis_info_clear(ref vi);
return(OV_EBADHEADER);
}
/* also responsible for range checking */
static vorbis_info_mapping mapping0_unpack(ref vorbis_info vi, ref Ogg.oggpack_buffer opb)
{
int i, b;
vorbis_info_mapping _info = new vorbis_info_mapping0();
vorbis_info_mapping0 info = _info as vorbis_info_mapping0;
codec_setup_info ci = vi.codec_setup as codec_setup_info;
b = Ogg.oggpack_read(ref opb, 1);
if (b < 0)
{
goto err_out;
}
if (b != 0)
{
info.submaps = Ogg.oggpack_read(ref opb, 4) + 1;
if (info.submaps <= 0)
{
goto err_out;
}
}
else
{
info.submaps = 1;
}
b = Ogg.oggpack_read(ref opb, 1);
if (b < 0)
{
goto err_out;
}
if (b != 0)
{
info.coupling_steps = Ogg.oggpack_read(ref opb, 8) + 1;
if (info.coupling_steps <= 0)
{
goto err_out;
}
for (i = 0; i < info.coupling_steps; i++)
{
int testM = info.coupling_mag[i] = Ogg.oggpack_read(ref opb, ilog((uint)vi.channels));
int testA = info.coupling_ang[i] = Ogg.oggpack_read(ref opb, ilog((uint)vi.channels));
if (testM < 0 || testA < 0 || testM == testA || testM >= vi.channels || testA >= vi.channels)
{
goto err_out;
}
}
}
if (Ogg.oggpack_read(ref opb, 2) != 0) /* 2,3:reserved */
{
goto err_out;
}
if (info.submaps > 1)
{
for (i = 0; i < vi.channels; i++)
{
info.chmuxlist[i] = Ogg.oggpack_read(ref opb, 4);
if (info.chmuxlist[i] >= info.submaps || info.chmuxlist[i] < 0)
{
goto err_out;
}
}
}
for (i = 0; i < info.submaps; i++)
{
Ogg.oggpack_read(ref opb, 8); /* time submap unused */
info.floorsubmap[i] = Ogg.oggpack_read(ref opb, 8);
if (info.floorsubmap[i] >= ci.floors || info.floorsubmap[i] < 0)
{
goto err_out;
}
info.residuesubmap[i] = Ogg.oggpack_read(ref opb, 8);
if (info.residuesubmap[i] >= ci.residues || info.residuesubmap[i] < 0)
{
goto err_out;
}
}
return(info);
err_out:
mapping0_free_info(ref _info);
return(null);
}
static int mapping0_forward(ref vorbis_block vb)
{
vorbis_dsp_state vd = vb.vd;
vorbis_info vi = vd.vi;
codec_setup_info ci = vi.codec_setup as codec_setup_info;
private_state b = vb.vd.backend_state as private_state;
vorbis_block_internal vbi = vb._internal as vorbis_block_internal;
int n = vb.pcmend;
int i, j, k;
int * nonzero = stackalloc int[vi.channels];
float **gmdct = (float **)_vorbis_block_alloc(ref vb, vi.channels * sizeof(float *));
int ** iwork = (int **)_vorbis_block_alloc(ref vb, vi.channels * sizeof(int *));
int *** floor_posts = (int ***)_vorbis_block_alloc(ref vb, vi.channels * sizeof(int **));
float global_ampmax = vbi.ampmax;
float *local_ampmax = stackalloc float[vi.channels];
int blocktype = vbi.blocktype;
int modenumber = vb.W;
vorbis_info_mapping0 info = ci.map_param[modenumber] as vorbis_info_mapping0;
vorbis_look_psy psy_look = b.psy[blocktype + (vb.W != 0 ? 2 : 0)];
vb.mode = modenumber;
for (i = 0; i < vi.channels; i++)
{
float scale = 4.0f / n;
float scale_dB;
float *pcm = vb.pcm[i];
float *logfft = pcm;
iwork[i] = (int *)_vorbis_block_alloc(ref vb, (n / 2) * sizeof(int));
gmdct[i] = (float *)_vorbis_block_alloc(ref vb, (n / 2) * sizeof(float));
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
scale_dB = todB(scale) + 0.345f;
/* window the PCM data */
_vorbis_apply_window(pcm, ref b.window, ref ci.blocksizes, vb.lW, vb.W, vb.nW);
/* transform the PCM data */
/* only MDCT right now.... */
mdct_forward(b.transform[vb.W][0] as mdct_lookup, pcm, gmdct[i]);
/* FFT yields more accurate tonal estimation (not phase sensitive) */
drft_forward(ref b.fft_look[vb.W], pcm);
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
logfft[0] = scale_dB + todB(*pcm) + 0.345f;
local_ampmax[i] = logfft[0];
for (j = 1; j < n - 1; j += 2)
{
float temp = pcm[j] * pcm[j] + pcm[j + 1] * pcm[j + 1];
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
temp = logfft[(j + 1) >> 1] = scale_dB + 0.5f * todB(temp) + 0.345f;
if (temp > local_ampmax[i])
{
local_ampmax[i] = temp;
}
}
if (local_ampmax[i] > 0.0f)
{
local_ampmax[i] = 0.0f;
}
if (local_ampmax[i] > global_ampmax)
{
global_ampmax = local_ampmax[i];
}
}
{
float *noise = (float *)_vorbis_block_alloc(ref vb, n / 2 * sizeof(float));
float *tone = (float *)_vorbis_block_alloc(ref vb, n / 2 * sizeof(float));
for (i = 0; i < vi.channels; i++)
{
/* the encoder setup assumes that all the modes used by any
* specific bitrate tweaking use the same floor */
int submap = info.chmuxlist[i];
/* the following makes things clearer to *me* anyway */
float *mdct = gmdct[i];
float *logfft = vb.pcm[i];
float *logmdct = logfft + n / 2;
float *logmask = logfft;
vb.mode = modenumber;
floor_posts[i] = (int **)_vorbis_block_alloc(ref vb, PACKETBLOBS * sizeof(int *));
ZeroMemory(floor_posts[i], sizeof(int *) * PACKETBLOBS);
for (j = 0; j < n / 2; j++)
{
/* + .345 is a hack; the original todB estimation used on IEEE 754 compliant machines had a bug that
* returned dB values about a third of a decibel too high. The bug was harmless because tunings
* implicitly took that into account. However, fixing the bug in the estimator requires changing all the tunings as well.
* For now, it's easier to sync things back up here, and recalibrate the tunings in the next major model upgrade. */
logmdct[j] = todB(mdct[j]) + 0.345f;
/* first step; noise masking. Not only does 'noise masking' give us curves from which we can decide how much resolution
* to give noise parts of the spectrum, it also implicitly hands us a tonality estimate (the larger the value in the
* 'noise_depth' vector, the more tonal that area is) */
_vp_noisemask(ref psy_look, logmdct, noise); /* noise does not have by-frequency offset bias applied yet */
/* second step: 'all the other crap'; all the stuff that isn't computed/fit for bitrate management goes in the second psy
* vector. This includes tone masking, peak limiting and ATH */
_vp_tonemask(ref psy_look, logfft, tone, global_ampmax, local_ampmax[i]);
/* third step; we offset the noise vectors, overlay tone masking. We then do a floor1-specific line fit. If we're
* performing bitrate management, the line fit is performed multiple times for up/down tweakage on demand. */
_vp_offset_and_mix(ref psy_look, noise, tone, 1, logmask, mdct, logmdct);
/* this algorithm is hardwired to floor 1 for now; abort out if we're *not* floor1. This won't happen unless someone has
* broken the encode setup lib. Guard it anyway. */
if (ci.floor_type[info.floorsubmap[submap]] != 1)
{
return(-1);
}
floor_posts[i][PACKETBLOBS / 2] = floor1_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, logmdct, logmask);
/* are we managing bitrate? If so, perform two more fits for later rate tweaking (fits represent hi/lo) */
if (vorbis_bitrate_managed(ref vb) != 0 && floor_posts[i][PACKETBLOBS / 2] != null)
{
/* higher rate by way of lower noise curve */
_vp_offset_and_mix(ref psy_look, noise, tone, 2, logmask, mdct, logmdct);
floor_posts[i][PACKETBLOBS - 1] = floor1_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, logmdct, logmask);
/* lower rate by way of higher noise curve */
_vp_offset_and_mix(ref psy_look, noise, tone, 0, logmask, mdct, logmdct);
floor_posts[i][0] = floor1_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, logmdct, logmask);
/* we also interpolate a range of intermediate curves for
* intermediate rates */
for (k = 1; k < PACKETBLOBS / 2; k++)
{
floor_posts[i][k] = floor1_interpolate_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, floor_posts[i][0], floor_posts[i][PACKETBLOBS / 2], k * 65536 / (PACKETBLOBS / 2));
}
for (k = PACKETBLOBS / 2 + 1; k < PACKETBLOBS - 1; k++)
{
floor_posts[i][k] = floor1_interpolate_fit(ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, floor_posts[i][PACKETBLOBS / 2], floor_posts[i][PACKETBLOBS - 1], (k - PACKETBLOBS / 2) * 65536 / (PACKETBLOBS / 2));
}
}
}
}
vbi.ampmax = global_ampmax;
/*
* the next phases are performed once for vbr-only and PACKETBLOB
* times for bitrate managed modes.
*
* 1) encode actual mode being used
* 2) encode the floor for each channel, compute coded mask curve/res
* 3) normalize and couple.
* 4) encode residue
* 5) save packet bytes to the packetblob vector
*/
/* iterate over the many masking curve fits we've created */
{
int **couple_bundle = stackalloc int *[vi.channels];
int * zerobundle = stackalloc int[vi.channels];
for (k = (vorbis_bitrate_managed(ref vb) != 0 ? 0 : PACKETBLOBS / 2); k <= (vorbis_bitrate_managed(ref vb) != 0 ? PACKETBLOBS - 1 : PACKETBLOBS / 2); k++)
{
Ogg.oggpack_buffer opb = vbi.packetblob[k];
/* start out our new packet blob with packet type and mode */
/* Encode the packet type */
Ogg.oggpack_write(ref opb, 0, 1);
/* Encode the modenumber */
/* Encode frame mode, pre,post windowsize, then dispatch */
Ogg.oggpack_write(ref opb, (uint)modenumber, b.modebits);
if (vb.W != 0)
{
Ogg.oggpack_write(ref opb, (uint)vb.lW, 1);
Ogg.oggpack_write(ref opb, (uint)vb.nW, 1);
}
/* encode floor, compute masking curve, sep out residue */
for (i = 0; i < vi.channels; i++)
{
int submap = info.chmuxlist[i];
int *ilogmask = iwork[i];
nonzero[i] = floor1_encode(ref opb, ref vb, b.flr[info.floorsubmap[submap]] as vorbis_look_floor1, floor_posts[i][k], ilogmask);
}
/* our iteration is now based on masking curve, not prequant and coupling. Only one prequant/coupling step */
/* quantize/couple */
/* incomplete implementation that assumes the tree is all depth one, or no tree at all */
_vp_couple_quantize_normalize(k, ci.psy_g_param, ref psy_look, info, gmdct, iwork, nonzero, ci.psy_g_param.sliding_lowpass[vb.W, k], vi.channels);
/* classify and encode by submap */
for (i = 0; i < info.submaps; i++)
{
int ch_in_bundle = 0;
int **classifications;
int resnum = info.residuesubmap[i];
for (j = 0; j < vi.channels; j++)
{
if (info.chmuxlist[j] == i)
{
zerobundle[ch_in_bundle] = 0;
if (nonzero[j] != 0)
{
zerobundle[ch_in_bundle] = 1;
}
couple_bundle[ch_in_bundle++] = iwork[j];
}
}
classifications = _residue_P[ci.residue_type[resnum]]._class(ref vb, b.residue[resnum], couple_bundle, zerobundle, ch_in_bundle);
ch_in_bundle = 0;
for (j = 0; j < vi.channels; j++)
{
if (info.chmuxlist[j] == i)
{
couple_bundle[ch_in_bundle++] = iwork[j];
}
}
_residue_P[ci.residue_type[resnum]].forward(ref opb, ref vb, b.residue[resnum], couple_bundle, zerobundle, ch_in_bundle, classifications, i);
}
/* ok, done encoding. Next protopacket. */
}
}
return(0);
}
}